1. Field of the Invention
The present invention generally relates to speech communication systems and, more particularly, to systems for digital speech coding.
2. Related Art
Communication systems include both wireline and wireless radio based systems. Wireless communication systems are electrically connected with the wireline based systems and communicate with the mobile communication devices using radio frequency (“RF”) communication. Currently, the radio frequencies available for communication in cellular systems, for example, are in the cellular frequency range centered around 900 MHz and in the personal communication services (“PCS”) frequency range centered around 1900 MHz. Data and voice transmissions within the wireless system have a bandwidth that consumes a portion of the radio frequency. Due to increased traffic arising from the expanding popularity of wireless communication devices, such as cellular telephones, it is desirable to reduce bandwidth of transmissions within the wireless systems.
Digital transmission in wireless radio communications is increasingly applied to both voice and data due to noise immunity, reliability, compactness or equipment and the ability to implement sophisticated signal processing functions using digital techniques. Digital transmission of speech signals involves the steps of sampling an analog speech waveform with an analog-to-digital converter, speech compression (encoding), transmission, speech decompression (decoding), digital-to-analog conversion, and playback into an earpiece or a speaker. The sampling of the analog speech waveform with the analog-to-digital converter creates a digital signal represented by a number of bits. The number of bits used in the digital signal to represent the analog speech waveform, however, requires a large portion of communication bandwidth. For example, a speech signal that is sampled at a rate of 8000 Hz (once every 0.125 ms), where each sample is represented by 16 bits, will result in a bit rate of 128,000 bits per second, or 128 Kbps.
Speech compression may be used to reduce the number of bits that represent the speech signal, thereby reducing the bandwidth needed for the transmission. However, speech compression may result in the degradation of the quality of decompressed speech. In general, a higher bit rate will result in a higher quality, while a lower bit rate will result in a lower quality.
One conventional approach to provide a higher quality speech at a lower average bit rate involves varying the degree of speech compression (i.e., varying the bit rate) depending on the part of the speech signal being compressed. Typically, parts of the speech signal for which adequate perceptual representation is more difficult (such as voiced speech, plosives, or voiced onsets) are coded and transmitted using a higher number of bits. Conversely, parts of the speech for which adequate perceptual representation is less difficult (such as unvoiced, or silence between words) are coded with a lower number of bits. The dissimilar coding rates can be attained, for example, with a variable bit rate coder having multiple codecs operating at different rates. As a result, the average bit rate for the speech signal will be relatively lower than would be the case for a fixed bit rate that provides speech of similar quality, leading to a reduction in the amount of bandwidth needed to transmit a speech signal. Although a lower bit rate is achieved through the use of variable rate coding, systems utilizing this approach remain inefficient. For example, the determination of which rate to use for coding a frame of the speech signal is often not correct, leading to situations where unvoiced or silence frames are coded at higher rates than frames containing actual voice activity.
Thus, there is an intense need in the art for systems and methods of speech coding that can reduce the amount of bandwidth required for speech signal transmission by achieving lower average bit rates, while maintaining high quality.